Mercurial > mplayer.hg
view libao2/ao_pcm.c @ 24590:2c238fa777ff
ao_alsa: Fix get_space() return values larger than buffersize
After a buffer underrun the ALSA get_space() function sometimes returned
values larger than the ao had set in ao_data.buffersize. Fix this by
replacing the old check against MAX_OUTBURST by one against
ao_data.buffersize. There should be no need for the MAX_OUTBURST check;
the current MPlayer side should no longer have any constant limit on the
amount of data an ao can buffer or request at once.
The get_space() values larger than ao_data.buffersize triggered errors
in audio decoding causing the current attempt to fill audio buffers to
be aborted. I'm not sure how often that caused behavior noticeably worse
then an underrun already is.
author | uau |
---|---|
date | Mon, 24 Sep 2007 21:49:58 +0000 |
parents | fa99b3d31d13 |
children | 7a6a7f47109f |
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#include "config.h" #include <stdio.h> #include <stdlib.h> #include <string.h> #include "libavutil/common.h" #include "mpbswap.h" #include "subopt-helper.h" #include "libaf/af_format.h" #include "audio_out.h" #include "audio_out_internal.h" #include "mp_msg.h" #include "help_mp.h" static ao_info_t info = { "RAW PCM/WAVE file writer audio output", "pcm", "Atmosfear", "" }; LIBAO_EXTERN(pcm) extern int vo_pts; static char *ao_outputfilename = NULL; static int ao_pcm_waveheader = 1; static int fast = 0; #define WAV_ID_RIFF 0x46464952 /* "RIFF" */ #define WAV_ID_WAVE 0x45564157 /* "WAVE" */ #define WAV_ID_FMT 0x20746d66 /* "fmt " */ #define WAV_ID_DATA 0x61746164 /* "data" */ #define WAV_ID_PCM 0x0001 struct WaveHeader { uint32_t riff; uint32_t file_length; uint32_t wave; uint32_t fmt; uint32_t fmt_length; uint16_t fmt_tag; uint16_t channels; uint32_t sample_rate; uint32_t bytes_per_second; uint16_t block_align; uint16_t bits; uint32_t data; uint32_t data_length; }; /* init with default values */ static struct WaveHeader wavhdr; static FILE *fp = NULL; // to set/get/query special features/parameters static int control(int cmd,void *arg){ return -1; } // open & setup audio device // return: 1=success 0=fail static int init(int rate,int channels,int format,int flags){ int bits; opt_t subopts[] = { {"waveheader", OPT_ARG_BOOL, &ao_pcm_waveheader, NULL}, {"file", OPT_ARG_MSTRZ, &ao_outputfilename, NULL}, {"fast", OPT_ARG_BOOL, &fast, NULL}, {NULL} }; // set defaults ao_pcm_waveheader = 1; if (subopt_parse(ao_subdevice, subopts) != 0) { return 0; } if (!ao_outputfilename){ ao_outputfilename = strdup(ao_pcm_waveheader?"audiodump.wav":"audiodump.pcm"); } /* bits is only equal to format if (format == 8) or (format == 16); this means that the following "if" is a kludge and should really be a switch to be correct in all cases */ bits=8; switch(format){ case AF_FORMAT_S8: format=AF_FORMAT_U8; case AF_FORMAT_U8: break; default: format=AF_FORMAT_S16_LE; bits=16; break; } ao_data.outburst = 65536; ao_data.buffersize= 2*65536; ao_data.channels=channels; ao_data.samplerate=rate; ao_data.format=format; ao_data.bps=channels*rate*(bits/8); wavhdr.riff = le2me_32(WAV_ID_RIFF); wavhdr.wave = le2me_32(WAV_ID_WAVE); wavhdr.fmt = le2me_32(WAV_ID_FMT); wavhdr.fmt_length = le2me_32(16); wavhdr.fmt_tag = le2me_16(WAV_ID_PCM); wavhdr.channels = le2me_16(ao_data.channels); wavhdr.sample_rate = le2me_32(ao_data.samplerate); wavhdr.bytes_per_second = le2me_32(ao_data.bps); wavhdr.bits = le2me_16(bits); wavhdr.block_align = le2me_16(ao_data.channels * (bits / 8)); wavhdr.data = le2me_32(WAV_ID_DATA); wavhdr.data_length=le2me_32(0x7ffff000); wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename, (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format)); mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo); fp = fopen(ao_outputfilename, "wb"); if(fp) { if(ao_pcm_waveheader){ /* Reserve space for wave header */ fwrite(&wavhdr,sizeof(wavhdr),1,fp); wavhdr.file_length=wavhdr.data_length=0; } return 1; } mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_PCM_CantOpenOutputFile, ao_outputfilename); return 0; } // close audio device static void uninit(int immed){ if(ao_pcm_waveheader && fseek(fp, 0, SEEK_SET) == 0){ /* Write wave header */ wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8; wavhdr.file_length = le2me_32(wavhdr.file_length); wavhdr.data_length = le2me_32(wavhdr.data_length); fwrite(&wavhdr,sizeof(wavhdr),1,fp); } fclose(fp); if (ao_outputfilename) free(ao_outputfilename); ao_outputfilename = NULL; } // stop playing and empty buffers (for seeking/pause) static void reset(void){ } // stop playing, keep buffers (for pause) static void audio_pause(void) { // for now, just call reset(); reset(); } // resume playing, after audio_pause() static void audio_resume(void) { } // return: how many bytes can be played without blocking static int get_space(void){ if(vo_pts) return ao_data.pts < vo_pts + fast * 30000 ? ao_data.outburst : 0; return ao_data.outburst; } // plays 'len' bytes of 'data' // it should round it down to outburst*n // return: number of bytes played static int play(void* data,int len,int flags){ // let libaf to do the conversion... #if 0 //#ifdef WORDS_BIGENDIAN if (ao_data.format == AFMT_S16_LE) { unsigned short *buffer = (unsigned short *) data; register int i; for(i = 0; i < len/2; ++i) { buffer[i] = le2me_16(buffer[i]); } } #endif //printf("PCM: Writing chunk!\n"); fwrite(data,len,1,fp); if(ao_pcm_waveheader) wavhdr.data_length += len; return len; } // return: delay in seconds between first and last sample in buffer static float get_delay(void){ return 0.0; }